similar to: Troubleshooting load issues

All the calls are using ulaw. The files that I am playing are gsm. I suppose doing a file convert with sox to .ulaw may help but it should be able to do 500 calls without an issue. Can it possibly be a bug? if not how do I profile which call(s) can be causing the spike? On Wed, Apr 22, 2020 at 2:21 PM Telium Technical Support <support at telium.io> wrote: > Could some calls be arriving

Try setting transcode_via_sln=no in /etc/asterisk/asterisk.conf and restart Asterisk. A reload will NOT apply the new value. Setting it to no seems to smooth out CPU usage on one of my servers. On 4/22/20 2:01 PM, Dovid Bender wrote: > Hi, > > I have an Asterisk box which has an IVR that plays random gsm files. The > box has SSD's and two CPU E5-2695 v2 cpus with 64GB ram.

Could some calls be arriving with a different codec? (Is transcoding causing the spikes)? Are you limiting codecs to match your audio files? From: asterisk-users [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Dovid Bender Sent: Wednesday, April 22, 2020 2:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject:

In PHP something like: $pid = pcntl_fork(); if ($pid != 0) { // we are the parent // do parent stuff exit; } // we are the child, detatch from terminal $sid = posix_setsid(); if ($sid < 0) { die; } // do child stuff On 04/19/2019 02:00 PM, Mark Wiater wrote: > On 4/19/2019 1:49 PM, Dovid Bender wrote: >> Mark, >> >> I am using PHP agi and when forking

It seems that what ever I set in my answer handler does not show up in the hangup handler. In order to do billing I can't rely on the g option where the caller hangs up the call. Looks like I can either use h or a hangup handler along with the shared function. On Tue, Dec 26, 2017 at 4:40 PM, Eric Wieling <ewieling at nyigc.com> wrote: > Don't use an 'h' extension, use

Hello, I'm quite sure this question has already be asked previously but before diving into it with a lab setup, I would like to re-ask here the thereafter question. I've got a bunch of very old Asterisk boxes (lastest Asterisk version is 1.6.1.X), all belonging to the same network, I would like to centralize on a single Asterisk instance on a brand new box. This instance will be powered

Why not use OpenSips/Kamailoo in between? Where you want 1.1.1.1 you pass it along as is. Where you want 2.2.2.2 change the sdp in opensips/kamailio On Thu, Oct 29, 2020 at 20:44 David Cunningham <dcunningham at voisonics.com> wrote: > Hello, > > Does anyone know a way with chan_sip to tell Asterisk to use a specific IP > address for its end of the communication for a specific

Glenn, I can't use MySQL as each node currently has MySQL however there is a lot of data that is stored locally on each box. I may have to take this route if I can't find something else but that would mean syncing all sorts of data that does not need to be synced. On Tue, Jul 30, 2019 at 10:03 PM Glenn Geller (VDOPh) <ggeller at vdo-ph.com> wrote: > Hi Dovid, > >

I have an AGI that can sometimes take time complete. I don't want the dialplan to be held up by the agi. Is there any way to call it and have Asterisk continue with the dialplan? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20190410/4c704231/attachment.html>

I am having major problems with the first run of Snom 360s that rolled out last month. I am working with the US vendor and they in turn are working with Snom but I wanted to see of anyone else was using these or having issues with them. Issues: Speakerphone/Hands Free volume spikes up and down during a call. You have to manually set the volume during every call. This makes it totally unusable.

Hi, I have a dial plan where I need to notify an external system when a call was answered and when the call hung up. In both requests the start time needs to be the same. My Dialplan looks something like this: [outbound] Exten => _X.,1,Dial(SIP/${EXTEN}@1.1.1.1,,U(call-answer-from-carrier)) Exten => h,1,NoOp(ANSWERED_TIME: ${ANSWEREDTIME} >>> DIAL_TIME: ${DIALEDTIME}

Hello, I'm load testing a new Asterisk 13 system (Debian Stretch, packaged asterisk). One system writes CDR though an ODBC connection to a local Postgres database over the LAN. When sending 50 new calls per second with SIPp, I'm seeing one system outputs : taskprocessor.c: The 'subm:cdr_engine-00000003' task processor queue reached 5000 scheduled tasks again. This [1] thread

I'm using BLACKLIST() to check numbers, which does not like leading + signs. I want to test if there is a plus sign, and then remove it. I tried: ; strip leading plus sign same => n, Verbose( callerid 0:1 is ${CALLERID(num):0:1} ) same => n,ExecIf($["${CALLERID(num):0:1}" = "+"]?Set(CALLERID(num) = ${CALLERID(num):1})

Steve, These are live streams of events so I can't simply rip the audio. As I mentioned at the end of my email putting in a sleep did help a bit however there are only so many streams Asterisk will grab nicely at once with out spiking the CPU. I also tinkered a bit with real time here is what I found: 1) If we have cachertclasses=no then Asterisk will only pull the stream if some one is

Anyone know of any equipment that I can use to connect a laptop running asterisk to a POTS line (RJ11) ? Regards, Dovid __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

Hi, Does anyone know of any projects that would allow you to use Redis in place of AstDB? By in place of I don't mean for what Asterisk needs but to store values. For instance for CNAM currently we need to use an AGI to connect to redis to pull CNAM. So in place of: Set(CALLERID(name)=${DB(CNAM/${CALLERID(num)})} it would be done with redis for example:

Hi, I am running several Asterisk boxes with realtime around the world. Does anyone have a recommendation for a "light" db that would work with Asterisk that would also allow replication between all sites (so if I add an entry to one box it will work with the rest)? TIA. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I've been getting regular CPU usage spikes(50%-80%), due to asterisk (according to top). I never noticed this on 1.4, and I have top running in the background pretty much all the time. In between those spikes Asterisk stays under 10% CPU usage (I have a transcoder card, which helps). It's very regular, never any missed spike, or any spike in between the regular spikes. I

Dovid Bender writes: > The tcpdump that you are running is on the Asterisk box or via port > mirroring? It's on the asterisk box itself. I've already replaced the network card - no change. Thanks, Roel > Regards, > > Dovid > > -----Original Message----- > From: Roel van Meer <roel at 1afa.com> > Sender: asterisk-users-bounces at

On Tue, Jun 26, 2018 at 7:28 PM, Doug Lytle <support at drdos.info> wrote: > On 06/26/2018 07:20 PM, Dovid Bender wrote: > > Doug, > > I tried that as well. Even with my dialplan looking like this: > > > > Ordering by includes works for me under Asterisk 11 and 13 > > What does the output of the below show? > > dialplan show from-external > >